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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
kjellanderb24317b2016-02-10 07:54:43 -08002 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00003 *
kjellanderb24317b2016-02-10 07:54:43 -08004 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00009 */
10
11// This file contains the PeerConnection interface as defined in
12// http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections.
13// Applications must use this interface to implement peerconnection.
14// PeerConnectionFactory class provides factory methods to create
15// peerconnection, mediastream and media tracks objects.
16//
17// The Following steps are needed to setup a typical call using Jsep.
18// 1. Create a PeerConnectionFactoryInterface. Check constructors for more
19// information about input parameters.
20// 2. Create a PeerConnection object. Provide a configuration string which
21// points either to stun or turn server to generate ICE candidates and provide
22// an object that implements the PeerConnectionObserver interface.
23// 3. Create local MediaStream and MediaTracks using the PeerConnectionFactory
24// and add it to PeerConnection by calling AddStream.
25// 4. Create an offer and serialize it and send it to the remote peer.
26// 5. Once an ice candidate have been found PeerConnection will call the
27// observer function OnIceCandidate. The candidates must also be serialized and
28// sent to the remote peer.
29// 6. Once an answer is received from the remote peer, call
30// SetLocalSessionDescription with the offer and SetRemoteSessionDescription
31// with the remote answer.
32// 7. Once a remote candidate is received from the remote peer, provide it to
33// the peerconnection by calling AddIceCandidate.
34
35
36// The Receiver of a call can decide to accept or reject the call.
37// This decision will be taken by the application not peerconnection.
38// If application decides to accept the call
39// 1. Create PeerConnectionFactoryInterface if it doesn't exist.
40// 2. Create a new PeerConnection.
41// 3. Provide the remote offer to the new PeerConnection object by calling
42// SetRemoteSessionDescription.
43// 4. Generate an answer to the remote offer by calling CreateAnswer and send it
44// back to the remote peer.
45// 5. Provide the local answer to the new PeerConnection by calling
46// SetLocalSessionDescription with the answer.
47// 6. Provide the remote ice candidates by calling AddIceCandidate.
48// 7. Once a candidate have been found PeerConnection will call the observer
49// function OnIceCandidate. Send these candidates to the remote peer.
50
Henrik Kjellander15583c12016-02-10 10:53:12 +010051#ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_
52#define WEBRTC_API_PEERCONNECTIONINTERFACE_H_
henrike@webrtc.org28e20752013-07-10 00:45:36 +000053
kwibergd1fe2812016-04-27 06:47:29 -070054#include <memory>
deadbeef3edec7c2016-12-10 11:44:26 -080055#include <ostream>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000056#include <string>
kwiberg0eb15ed2015-12-17 03:04:15 -080057#include <utility>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000058#include <vector>
59
Henrik Kjellander15583c12016-02-10 10:53:12 +010060#include "webrtc/api/datachannelinterface.h"
Henrik Kjellander15583c12016-02-10 10:53:12 +010061#include "webrtc/api/dtmfsenderinterface.h"
62#include "webrtc/api/jsep.h"
63#include "webrtc/api/mediastreaminterface.h"
hbos74e1a4f2016-09-15 23:33:01 -070064#include "webrtc/api/rtcstatscollector.h"
Henrik Kjellander15583c12016-02-10 10:53:12 +010065#include "webrtc/api/rtpreceiverinterface.h"
66#include "webrtc/api/rtpsenderinterface.h"
67#include "webrtc/api/statstypes.h"
68#include "webrtc/api/umametrics.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000069#include "webrtc/base/fileutils.h"
phoglund@webrtc.org006521d2015-02-12 09:23:59 +000070#include "webrtc/base/network.h"
Henrik Boström87713d02015-08-25 09:53:21 +020071#include "webrtc/base/rtccertificate.h"
Henrik Boströmd03c23b2016-06-01 11:44:18 +020072#include "webrtc/base/rtccertificategenerator.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000073#include "webrtc/base/socketaddress.h"
kjellandera96e2d72016-02-04 23:52:28 -080074#include "webrtc/base/sslstreamadapter.h"
nissec36b31b2016-04-11 23:25:29 -070075#include "webrtc/media/base/mediachannel.h"
deadbeef41b07982015-12-01 15:01:24 -080076#include "webrtc/p2p/base/portallocator.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000077
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000078namespace rtc {
jiayl@webrtc.org61e00b02015-03-04 22:17:38 +000079class SSLIdentity;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000080class Thread;
81}
82
83namespace cricket {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000084class WebRtcVideoDecoderFactory;
85class WebRtcVideoEncoderFactory;
86}
87
88namespace webrtc {
89class AudioDeviceModule;
gyzhou95aa9642016-12-13 14:06:26 -080090class AudioMixer;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000091class MediaConstraintsInterface;
92
93// MediaStream container interface.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000094class StreamCollectionInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000095 public:
96 // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
97 virtual size_t count() = 0;
98 virtual MediaStreamInterface* at(size_t index) = 0;
99 virtual MediaStreamInterface* find(const std::string& label) = 0;
100 virtual MediaStreamTrackInterface* FindAudioTrack(
101 const std::string& id) = 0;
102 virtual MediaStreamTrackInterface* FindVideoTrack(
103 const std::string& id) = 0;
104
105 protected:
106 // Dtor protected as objects shouldn't be deleted via this interface.
107 ~StreamCollectionInterface() {}
108};
109
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000110class StatsObserver : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000111 public:
nisseb36ee8d2016-12-20 03:30:00 -0800112 // TODO(nisse, hbos): Old version, not passing ownership. Should
113 // perhaps be deprecated, but since all of this is a legacy
114 // interface anyway, probably best to leave as is until this class
115 // can be deleted.
116 virtual void OnComplete(const StatsReports& reports) {}
117 virtual void OnCompleteReports(std::unique_ptr<StatsReports> reports) {
118 OnComplete(*reports);
119 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000120
121 protected:
122 virtual ~StatsObserver() {}
123};
124
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000125class MetricsObserverInterface : public rtc::RefCountInterface {
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000126 public:
Guo-wei Shieh3d564c12015-08-19 16:51:15 -0700127
128 // |type| is the type of the enum counter to be incremented. |counter|
129 // is the particular counter in that type. |counter_max| is the next sequence
130 // number after the highest counter.
131 virtual void IncrementEnumCounter(PeerConnectionEnumCounterType type,
132 int counter,
133 int counter_max) {}
134
Guo-wei Shieh456696a2015-09-30 21:48:54 -0700135 // This is used to handle sparse counters like SSL cipher suites.
136 // TODO(guoweis): Remove the implementation once the dependency's interface
137 // definition is updated.
138 virtual void IncrementSparseEnumCounter(PeerConnectionEnumCounterType type,
139 int counter) {
140 IncrementEnumCounter(type, counter, 0 /* Ignored */);
141 }
142
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000143 virtual void AddHistogramSample(PeerConnectionMetricsName type,
mallinath@webrtc.orgd37bcfa2014-05-12 23:10:18 +0000144 int value) = 0;
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000145
146 protected:
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000147 virtual ~MetricsObserverInterface() {}
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000148};
149
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000150typedef MetricsObserverInterface UMAObserver;
151
deadbeef3edec7c2016-12-10 11:44:26 -0800152// Enumeration to represent distinct classes of errors that an application
deadbeef1e234612016-12-24 01:43:32 -0800153// may wish to act upon differently. These roughly map to DOMExceptions in
154// the web API, as described in the comments below.
155enum class RtcError {
deadbeef3edec7c2016-12-10 11:44:26 -0800156 // No error.
157 NONE,
158 // A supplied parameter is valid, but currently unsupported.
159 // Maps to InvalidAccessError DOMException.
160 UNSUPPORTED_PARAMETER,
161 // General error indicating that a supplied parameter is invalid.
162 // Maps to InvalidAccessError or TypeError DOMException depending on context.
163 INVALID_PARAMETER,
164 // Slightly more specific than INVALID_PARAMETER; a parameter's value was
165 // outside the allowed range.
166 // Maps to RangeError DOMException.
167 INVALID_RANGE,
168 // Slightly more specific than INVALID_PARAMETER; an error occurred while
169 // parsing string input.
170 // Maps to SyntaxError DOMException.
171 SYNTAX_ERROR,
172 // The object does not support this operation in its current state.
173 // Maps to InvalidStateError DOMException.
174 INVALID_STATE,
175 // An attempt was made to modify the object in an invalid way.
176 // Maps to InvalidModificationError DOMException.
177 INVALID_MODIFICATION,
178 // An error occurred within an underlying network protocol.
179 // Maps to NetworkError DOMException.
180 NETWORK_ERROR,
181 // The operation failed due to an internal error.
182 // Maps to OperationError DOMException.
183 INTERNAL_ERROR,
184};
185
186// Outputs the error as a friendly string.
187// Update this method when adding a new error type.
deadbeef1e234612016-12-24 01:43:32 -0800188std::ostream& operator<<(std::ostream& stream, RtcError error);
deadbeef3edec7c2016-12-10 11:44:26 -0800189
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000190class PeerConnectionInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000191 public:
192 // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
193 enum SignalingState {
194 kStable,
195 kHaveLocalOffer,
196 kHaveLocalPrAnswer,
197 kHaveRemoteOffer,
198 kHaveRemotePrAnswer,
199 kClosed,
200 };
201
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000202 enum IceGatheringState {
203 kIceGatheringNew,
204 kIceGatheringGathering,
205 kIceGatheringComplete
206 };
207
208 enum IceConnectionState {
209 kIceConnectionNew,
210 kIceConnectionChecking,
211 kIceConnectionConnected,
212 kIceConnectionCompleted,
213 kIceConnectionFailed,
214 kIceConnectionDisconnected,
215 kIceConnectionClosed,
Guo-wei Shieh3d564c12015-08-19 16:51:15 -0700216 kIceConnectionMax,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000217 };
218
219 struct IceServer {
Joachim Bauch7c4e7452015-05-28 23:06:30 +0200220 // TODO(jbauch): Remove uri when all code using it has switched to urls.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000221 std::string uri;
Joachim Bauch7c4e7452015-05-28 23:06:30 +0200222 std::vector<std::string> urls;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000223 std::string username;
224 std::string password;
deadbeefd1a38b52016-12-10 13:15:33 -0800225 bool operator==(const IceServer& o) const {
226 return uri == o.uri && urls == o.urls && username == o.username &&
magjedd5236e22016-12-20 02:22:06 -0800227 password == o.password;
deadbeefd1a38b52016-12-10 13:15:33 -0800228 }
229 bool operator!=(const IceServer& o) const { return !(*this == o); }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000230 };
231 typedef std::vector<IceServer> IceServers;
232
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000233 enum IceTransportsType {
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000234 // TODO(pthatcher): Rename these kTransporTypeXXX, but update
235 // Chromium at the same time.
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000236 kNone,
237 kRelay,
238 kNoHost,
239 kAll
240 };
241
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000242 // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-08#section-4.1.1
243 enum BundlePolicy {
244 kBundlePolicyBalanced,
245 kBundlePolicyMaxBundle,
246 kBundlePolicyMaxCompat
247 };
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000248
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700249 // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
250 enum RtcpMuxPolicy {
251 kRtcpMuxPolicyNegotiate,
252 kRtcpMuxPolicyRequire,
253 };
254
Jiayang Liucac1b382015-04-30 12:35:24 -0700255 enum TcpCandidatePolicy {
256 kTcpCandidatePolicyEnabled,
257 kTcpCandidatePolicyDisabled
258 };
259
honghaiz60347052016-05-31 18:29:12 -0700260 enum CandidateNetworkPolicy {
261 kCandidateNetworkPolicyAll,
262 kCandidateNetworkPolicyLowCost
263 };
264
honghaiz1f429e32015-09-28 07:57:34 -0700265 enum ContinualGatheringPolicy {
266 GATHER_ONCE,
267 GATHER_CONTINUALLY
268 };
269
Honghai Zhangf7ddc062016-09-01 15:34:01 -0700270 enum class RTCConfigurationType {
271 // A configuration that is safer to use, despite not having the best
272 // performance. Currently this is the default configuration.
273 kSafe,
274 // An aggressive configuration that has better performance, although it
275 // may be riskier and may need extra support in the application.
276 kAggressive
277 };
278
Henrik Boström87713d02015-08-25 09:53:21 +0200279 // TODO(hbos): Change into class with private data and public getters.
nissec36b31b2016-04-11 23:25:29 -0700280 // TODO(nisse): In particular, accessing fields directly from an
281 // application is brittle, since the organization mirrors the
282 // organization of the implementation, which isn't stable. So we
283 // need getters and setters at least for fields which applications
284 // are interested in.
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000285 struct RTCConfiguration {
Niels Möller71bdda02016-03-31 12:59:59 +0200286 // This struct is subject to reorganization, both for naming
287 // consistency, and to group settings to match where they are used
288 // in the implementation. To do that, we need getter and setter
289 // methods for all settings which are of interest to applications,
290 // Chrome in particular.
291
Honghai Zhangf7ddc062016-09-01 15:34:01 -0700292 RTCConfiguration() = default;
293 RTCConfiguration(RTCConfigurationType type) {
294 if (type == RTCConfigurationType::kAggressive) {
Honghai Zhangaecd9822016-09-02 16:58:17 -0700295 // These parameters are also defined in Java and IOS configurations,
296 // so their values may be overwritten by the Java or IOS configuration.
297 bundle_policy = kBundlePolicyMaxBundle;
298 rtcp_mux_policy = kRtcpMuxPolicyRequire;
299 ice_connection_receiving_timeout =
300 kAggressiveIceConnectionReceivingTimeout;
301
302 // These parameters are not defined in Java or IOS configuration,
303 // so their values will not be overwritten.
304 enable_ice_renomination = true;
Honghai Zhangf7ddc062016-09-01 15:34:01 -0700305 redetermine_role_on_ice_restart = false;
306 }
Honghai Zhangbfd398c2016-08-30 22:07:42 -0700307 }
308
nissec36b31b2016-04-11 23:25:29 -0700309 bool dscp() { return media_config.enable_dscp; }
310 void set_dscp(bool enable) { media_config.enable_dscp = enable; }
Niels Möller71bdda02016-03-31 12:59:59 +0200311
312 // TODO(nisse): The corresponding flag in MediaConfig and
313 // elsewhere should be renamed enable_cpu_adaptation.
nissec36b31b2016-04-11 23:25:29 -0700314 bool cpu_adaptation() {
315 return media_config.video.enable_cpu_overuse_detection;
316 }
Niels Möller71bdda02016-03-31 12:59:59 +0200317 void set_cpu_adaptation(bool enable) {
nissec36b31b2016-04-11 23:25:29 -0700318 media_config.video.enable_cpu_overuse_detection = enable;
Niels Möller71bdda02016-03-31 12:59:59 +0200319 }
320
nissec36b31b2016-04-11 23:25:29 -0700321 bool suspend_below_min_bitrate() {
322 return media_config.video.suspend_below_min_bitrate;
323 }
Niels Möller71bdda02016-03-31 12:59:59 +0200324 void set_suspend_below_min_bitrate(bool enable) {
nissec36b31b2016-04-11 23:25:29 -0700325 media_config.video.suspend_below_min_bitrate = enable;
Niels Möller71bdda02016-03-31 12:59:59 +0200326 }
327
328 // TODO(nisse): The negation in the corresponding MediaConfig
329 // attribute is inconsistent, and it should be renamed at some
330 // point.
nissec36b31b2016-04-11 23:25:29 -0700331 bool prerenderer_smoothing() {
332 return !media_config.video.disable_prerenderer_smoothing;
333 }
Niels Möller71bdda02016-03-31 12:59:59 +0200334 void set_prerenderer_smoothing(bool enable) {
nissec36b31b2016-04-11 23:25:29 -0700335 media_config.video.disable_prerenderer_smoothing = !enable;
Niels Möller71bdda02016-03-31 12:59:59 +0200336 }
337
honghaiz4edc39c2015-09-01 09:53:56 -0700338 static const int kUndefined = -1;
339 // Default maximum number of packets in the audio jitter buffer.
340 static const int kAudioJitterBufferMaxPackets = 50;
Honghai Zhangaecd9822016-09-02 16:58:17 -0700341 // ICE connection receiving timeout for aggressive configuration.
342 static const int kAggressiveIceConnectionReceivingTimeout = 1000;
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000343 // TODO(pthatcher): Rename this ice_transport_type, but update
344 // Chromium at the same time.
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700345 IceTransportsType type = kAll;
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000346 // TODO(pthatcher): Rename this ice_servers, but update Chromium
347 // at the same time.
348 IceServers servers;
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700349 BundlePolicy bundle_policy = kBundlePolicyBalanced;
zhihuang4dfb8ce2016-11-23 10:30:12 -0800350 RtcpMuxPolicy rtcp_mux_policy = kRtcpMuxPolicyRequire;
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700351 TcpCandidatePolicy tcp_candidate_policy = kTcpCandidatePolicyEnabled;
honghaiz60347052016-05-31 18:29:12 -0700352 CandidateNetworkPolicy candidate_network_policy =
353 kCandidateNetworkPolicyAll;
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700354 int audio_jitter_buffer_max_packets = kAudioJitterBufferMaxPackets;
355 bool audio_jitter_buffer_fast_accelerate = false;
356 int ice_connection_receiving_timeout = kUndefined; // ms
357 int ice_backup_candidate_pair_ping_interval = kUndefined; // ms
358 ContinualGatheringPolicy continual_gathering_policy = GATHER_ONCE;
Henrik Boström87713d02015-08-25 09:53:21 +0200359 std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700360 bool prioritize_most_likely_ice_candidate_pairs = false;
nissec36b31b2016-04-11 23:25:29 -0700361 struct cricket::MediaConfig media_config;
htaa2a49d92016-03-04 02:51:39 -0800362 // Flags corresponding to values set by constraint flags.
363 // rtc::Optional flags can be "missing", in which case the webrtc
364 // default applies.
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700365 bool disable_ipv6 = false;
366 bool enable_rtp_data_channel = false;
zhihuang9763d562016-08-05 11:14:50 -0700367 bool enable_quic = false;
htaa2a49d92016-03-04 02:51:39 -0800368 rtc::Optional<int> screencast_min_bitrate;
369 rtc::Optional<bool> combined_audio_video_bwe;
370 rtc::Optional<bool> enable_dtls_srtp;
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700371 int ice_candidate_pool_size = 0;
Honghai Zhangb9e7b4a2016-06-30 20:52:02 -0700372 bool prune_turn_ports = false;
Taylor Brandstettere9851112016-07-01 11:11:13 -0700373 // If set to true, this means the ICE transport should presume TURN-to-TURN
374 // candidate pairs will succeed, even before a binding response is received.
375 bool presume_writable_when_fully_relayed = false;
Honghai Zhang4cedf2b2016-08-31 08:18:11 -0700376 // If true, "renomination" will be added to the ice options in the transport
377 // description.
378 bool enable_ice_renomination = false;
Honghai Zhangbfd398c2016-08-30 22:07:42 -0700379 // If true, ICE role is redetermined when peerconnection sets a local
380 // transport description that indicates an ICE restart.
381 bool redetermine_role_on_ice_restart = true;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000382 };
383
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000384 struct RTCOfferAnswerOptions {
385 static const int kUndefined = -1;
386 static const int kMaxOfferToReceiveMedia = 1;
387
388 // The default value for constraint offerToReceiveX:true.
389 static const int kOfferToReceiveMediaTrue = 1;
390
Honghai Zhang4cedf2b2016-08-31 08:18:11 -0700391 int offer_to_receive_video = kUndefined;
392 int offer_to_receive_audio = kUndefined;
393 bool voice_activity_detection = true;
394 bool ice_restart = false;
395 bool use_rtp_mux = true;
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000396
Honghai Zhang4cedf2b2016-08-31 08:18:11 -0700397 RTCOfferAnswerOptions() = default;
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000398
399 RTCOfferAnswerOptions(int offer_to_receive_video,
400 int offer_to_receive_audio,
401 bool voice_activity_detection,
402 bool ice_restart,
403 bool use_rtp_mux)
404 : offer_to_receive_video(offer_to_receive_video),
405 offer_to_receive_audio(offer_to_receive_audio),
406 voice_activity_detection(voice_activity_detection),
407 ice_restart(ice_restart),
408 use_rtp_mux(use_rtp_mux) {}
409 };
410
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000411 // Used by GetStats to decide which stats to include in the stats reports.
412 // |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
413 // |kStatsOutputLevelDebug| includes both the standard stats and additional
414 // stats for debugging purposes.
415 enum StatsOutputLevel {
416 kStatsOutputLevelStandard,
417 kStatsOutputLevelDebug,
418 };
419
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000420 // Accessor methods to active local streams.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000421 virtual rtc::scoped_refptr<StreamCollectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000422 local_streams() = 0;
423
424 // Accessor methods to remote streams.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000425 virtual rtc::scoped_refptr<StreamCollectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000426 remote_streams() = 0;
427
428 // Add a new MediaStream to be sent on this PeerConnection.
429 // Note that a SessionDescription negotiation is needed before the
430 // remote peer can receive the stream.
perkj@webrtc.orgfd0efb62014-11-06 12:16:36 +0000431 virtual bool AddStream(MediaStreamInterface* stream) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000432
433 // Remove a MediaStream from this PeerConnection.
434 // Note that a SessionDescription negotiation is need before the
435 // remote peer is notified.
436 virtual void RemoveStream(MediaStreamInterface* stream) = 0;
437
deadbeefe1f9d832016-01-14 15:35:42 -0800438 // TODO(deadbeef): Make the following two methods pure virtual once
439 // implemented by all subclasses of PeerConnectionInterface.
440 // Add a new MediaStreamTrack to be sent on this PeerConnection.
441 // |streams| indicates which stream labels the track should be associated
442 // with.
443 virtual rtc::scoped_refptr<RtpSenderInterface> AddTrack(
444 MediaStreamTrackInterface* track,
445 std::vector<MediaStreamInterface*> streams) {
446 return nullptr;
447 }
448
449 // Remove an RtpSender from this PeerConnection.
450 // Returns true on success.
451 virtual bool RemoveTrack(RtpSenderInterface* sender) {
452 return false;
453 }
454
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000455 // Returns pointer to the created DtmfSender on success.
456 // Otherwise returns NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000457 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000458 AudioTrackInterface* track) = 0;
459
deadbeef70ab1a12015-09-28 16:53:55 -0700460 // TODO(deadbeef): Make these pure virtual once all subclasses implement them.
deadbeeffac06552015-11-25 11:26:01 -0800461 // |kind| must be "audio" or "video".
deadbeefbd7d8f72015-12-18 16:58:44 -0800462 // |stream_id| is used to populate the msid attribute; if empty, one will
463 // be generated automatically.
deadbeeffac06552015-11-25 11:26:01 -0800464 virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
deadbeefbd7d8f72015-12-18 16:58:44 -0800465 const std::string& kind,
466 const std::string& stream_id) {
deadbeeffac06552015-11-25 11:26:01 -0800467 return rtc::scoped_refptr<RtpSenderInterface>();
468 }
469
deadbeef70ab1a12015-09-28 16:53:55 -0700470 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
471 const {
472 return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
473 }
474
475 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
476 const {
477 return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
478 }
479
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000480 virtual bool GetStats(StatsObserver* observer,
481 MediaStreamTrackInterface* track,
482 StatsOutputLevel level) = 0;
hbos74e1a4f2016-09-15 23:33:01 -0700483 // Gets stats using the new stats collection API, see webrtc/api/stats/. These
484 // will replace old stats collection API when the new API has matured enough.
hbose3810152016-12-13 02:35:19 -0800485 // TODO(hbos): Default implementation that does nothing only exists as to not
486 // break third party projects. As soon as they have been updated this should
487 // be changed to "= 0;".
488 virtual void GetStats(RTCStatsCollectorCallback* callback) {}
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000489
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000490 virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000491 const std::string& label,
492 const DataChannelInit* config) = 0;
493
494 virtual const SessionDescriptionInterface* local_description() const = 0;
495 virtual const SessionDescriptionInterface* remote_description() const = 0;
deadbeeffe4a8a42016-12-20 17:56:17 -0800496 // A "current" description the one currently negotiated from a complete
497 // offer/answer exchange.
498 virtual const SessionDescriptionInterface* current_local_description() const {
499 return nullptr;
500 }
501 virtual const SessionDescriptionInterface* current_remote_description()
502 const {
503 return nullptr;
504 }
505 // A "pending" description is one that's part of an incomplete offer/answer
506 // exchange (thus, either an offer or a pranswer). Once the offer/answer
507 // exchange is finished, the "pending" description will become "current".
508 virtual const SessionDescriptionInterface* pending_local_description() const {
509 return nullptr;
510 }
511 virtual const SessionDescriptionInterface* pending_remote_description()
512 const {
513 return nullptr;
514 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000515
516 // Create a new offer.
517 // The CreateSessionDescriptionObserver callback will be called when done.
518 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000519 const MediaConstraintsInterface* constraints) {}
520
521 // TODO(jiayl): remove the default impl and the old interface when chromium
522 // code is updated.
523 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
524 const RTCOfferAnswerOptions& options) {}
525
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000526 // Create an answer to an offer.
527 // The CreateSessionDescriptionObserver callback will be called when done.
528 virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
htaa2a49d92016-03-04 02:51:39 -0800529 const RTCOfferAnswerOptions& options) {}
530 // Deprecated - use version above.
531 // TODO(hta): Remove and remove default implementations when all callers
532 // are updated.
533 virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
534 const MediaConstraintsInterface* constraints) {}
535
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000536 // Sets the local session description.
537 // JsepInterface takes the ownership of |desc| even if it fails.
538 // The |observer| callback will be called when done.
539 virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
540 SessionDescriptionInterface* desc) = 0;
541 // Sets the remote session description.
542 // JsepInterface takes the ownership of |desc| even if it fails.
543 // The |observer| callback will be called when done.
544 virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
545 SessionDescriptionInterface* desc) = 0;
546 // Restarts or updates the ICE Agent process of gathering local candidates
547 // and pinging remote candidates.
deadbeefa67696b2015-09-29 11:56:26 -0700548 // TODO(deadbeef): Remove once Chrome is moved over to SetConfiguration.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000549 virtual bool UpdateIce(const IceServers& configuration,
deadbeefa67696b2015-09-29 11:56:26 -0700550 const MediaConstraintsInterface* constraints) {
551 return false;
552 }
htaa2a49d92016-03-04 02:51:39 -0800553 virtual bool UpdateIce(const IceServers& configuration) { return false; }
deadbeef46c73892016-11-16 19:42:04 -0800554 // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
555 // PeerConnectionInterface implement it.
556 virtual PeerConnectionInterface::RTCConfiguration GetConfiguration() {
557 return PeerConnectionInterface::RTCConfiguration();
558 }
deadbeefa67696b2015-09-29 11:56:26 -0700559 // Sets the PeerConnection's global configuration to |config|.
560 // Any changes to STUN/TURN servers or ICE candidate policy will affect the
561 // next gathering phase, and cause the next call to createOffer to generate
deadbeef1e234612016-12-24 01:43:32 -0800562 // new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
563 // cannot be changed with this method.
deadbeefa67696b2015-09-29 11:56:26 -0700564 // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
565 // PeerConnectionInterface implement it.
566 virtual bool SetConfiguration(
deadbeef1e234612016-12-24 01:43:32 -0800567 const PeerConnectionInterface::RTCConfiguration& config) {
deadbeefa67696b2015-09-29 11:56:26 -0700568 return false;
569 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000570 // Provides a remote candidate to the ICE Agent.
571 // A copy of the |candidate| will be created and added to the remote
572 // description. So the caller of this method still has the ownership of the
573 // |candidate|.
574 // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
575 // take the ownership of the |candidate|.
576 virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
577
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700578 // Removes a group of remote candidates from the ICE agent.
579 virtual bool RemoveIceCandidates(
580 const std::vector<cricket::Candidate>& candidates) {
581 return false;
582 }
583
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000584 virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
585
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000586 // Returns the current SignalingState.
587 virtual SignalingState signaling_state() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000588 virtual IceConnectionState ice_connection_state() = 0;
589 virtual IceGatheringState ice_gathering_state() = 0;
590
ivoc14d5dbe2016-07-04 07:06:55 -0700591 // Starts RtcEventLog using existing file. Takes ownership of |file| and
592 // passes it on to Call, which will take the ownership. If the
593 // operation fails the file will be closed. The logging will stop
594 // automatically after 10 minutes have passed, or when the StopRtcEventLog
595 // function is called.
596 // TODO(ivoc): Make this pure virtual when Chrome is updated.
597 virtual bool StartRtcEventLog(rtc::PlatformFile file,
598 int64_t max_size_bytes) {
599 return false;
600 }
601
602 // Stops logging the RtcEventLog.
603 // TODO(ivoc): Make this pure virtual when Chrome is updated.
604 virtual void StopRtcEventLog() {}
605
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000606 // Terminates all media and closes the transport.
607 virtual void Close() = 0;
608
609 protected:
610 // Dtor protected as objects shouldn't be deleted via this interface.
611 ~PeerConnectionInterface() {}
612};
613
614// PeerConnection callback interface. Application should implement these
615// methods.
616class PeerConnectionObserver {
617 public:
618 enum StateType {
619 kSignalingState,
620 kIceState,
621 };
622
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000623 // Triggered when the SignalingState changed.
624 virtual void OnSignalingChange(
perkjdfb769d2016-02-09 03:09:43 -0800625 PeerConnectionInterface::SignalingState new_state) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000626
Taylor Brandstetter98cde262016-05-31 13:02:21 -0700627 // TODO(deadbeef): Once all subclasses override the scoped_refptr versions
628 // of the below three methods, make them pure virtual and remove the raw
629 // pointer version.
630
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000631 // Triggered when media is received on a new stream from remote peer.
Taylor Brandstetter98cde262016-05-31 13:02:21 -0700632 virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}
633 // Deprecated; please use the version that uses a scoped_refptr.
634 virtual void OnAddStream(MediaStreamInterface* stream) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000635
636 // Triggered when a remote peer close a stream.
Taylor Brandstetter98cde262016-05-31 13:02:21 -0700637 virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) {
638 }
639 // Deprecated; please use the version that uses a scoped_refptr.
640 virtual void OnRemoveStream(MediaStreamInterface* stream) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000641
Taylor Brandstetter98cde262016-05-31 13:02:21 -0700642 // Triggered when a remote peer opens a data channel.
643 virtual void OnDataChannel(
644 rtc::scoped_refptr<DataChannelInterface> data_channel){};
645 // Deprecated; please use the version that uses a scoped_refptr.
646 virtual void OnDataChannel(DataChannelInterface* data_channel) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000647
Taylor Brandstetter98cde262016-05-31 13:02:21 -0700648 // Triggered when renegotiation is needed. For example, an ICE restart
649 // has begun.
fischman@webrtc.orgd7568a02014-01-13 22:04:12 +0000650 virtual void OnRenegotiationNeeded() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000651
Taylor Brandstetter98cde262016-05-31 13:02:21 -0700652 // Called any time the IceConnectionState changes.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000653 virtual void OnIceConnectionChange(
perkjdfb769d2016-02-09 03:09:43 -0800654 PeerConnectionInterface::IceConnectionState new_state) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000655
Taylor Brandstetter98cde262016-05-31 13:02:21 -0700656 // Called any time the IceGatheringState changes.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000657 virtual void OnIceGatheringChange(
perkjdfb769d2016-02-09 03:09:43 -0800658 PeerConnectionInterface::IceGatheringState new_state) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000659
Taylor Brandstetter98cde262016-05-31 13:02:21 -0700660 // A new ICE candidate has been gathered.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000661 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
662
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700663 // Ice candidates have been removed.
664 // TODO(honghaiz): Make this a pure virtual method when all its subclasses
665 // implement it.
666 virtual void OnIceCandidatesRemoved(
667 const std::vector<cricket::Candidate>& candidates) {}
668
Peter Thatcher54360512015-07-08 11:08:35 -0700669 // Called when the ICE connection receiving status changes.
670 virtual void OnIceConnectionReceivingChange(bool receiving) {}
671
zhihuang81c3a032016-11-17 12:06:24 -0800672 // Called when a track is added to streams.
673 // TODO(zhihuang) Make this a pure virtual method when all its subclasses
674 // implement it.
675 virtual void OnAddTrack(
676 rtc::scoped_refptr<RtpReceiverInterface> receiver,
zhihuangc63b8942016-12-02 15:41:10 -0800677 const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {}
zhihuang81c3a032016-11-17 12:06:24 -0800678
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000679 protected:
680 // Dtor protected as objects shouldn't be deleted via this interface.
681 ~PeerConnectionObserver() {}
682};
683
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000684// PeerConnectionFactoryInterface is the factory interface use for creating
685// PeerConnection, MediaStream and media tracks.
686// PeerConnectionFactoryInterface will create required libjingle threads,
687// socket and network manager factory classes for networking.
688// If an application decides to provide its own threads and network
689// implementation of these classes it should use the alternate
690// CreatePeerConnectionFactory method which accepts threads as input and use the
Taylor Brandstetter0c7e9f52015-12-29 14:14:52 -0800691// CreatePeerConnection version that takes a PortAllocator as an
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000692// argument.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000693class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000694 public:
wu@webrtc.org97077a32013-10-25 21:18:33 +0000695 class Options {
696 public:
Guo-wei Shieha7446d22016-01-11 15:27:03 -0800697 Options()
698 : disable_encryption(false),
699 disable_sctp_data_channels(false),
700 disable_network_monitor(false),
701 network_ignore_mask(rtc::kDefaultNetworkIgnoreMask),
jbauchcb560652016-08-04 05:20:32 -0700702 ssl_max_version(rtc::SSL_PROTOCOL_DTLS_12),
703 crypto_options(rtc::CryptoOptions::NoGcm()) {}
wu@webrtc.org97077a32013-10-25 21:18:33 +0000704 bool disable_encryption;
705 bool disable_sctp_data_channels;
honghaiz023f3ef2015-10-19 09:39:32 -0700706 bool disable_network_monitor;
phoglund@webrtc.org006521d2015-02-12 09:23:59 +0000707
708 // Sets the network types to ignore. For instance, calling this with
709 // ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
710 // loopback interfaces.
711 int network_ignore_mask;
Joachim Bauch04e5b492015-05-29 09:40:39 +0200712
713 // Sets the maximum supported protocol version. The highest version
714 // supported by both ends will be used for the connection, i.e. if one
715 // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
716 rtc::SSLProtocolVersion ssl_max_version;
jbauchcb560652016-08-04 05:20:32 -0700717
718 // Sets crypto related options, e.g. enabled cipher suites.
719 rtc::CryptoOptions crypto_options;
wu@webrtc.org97077a32013-10-25 21:18:33 +0000720 };
721
722 virtual void SetOptions(const Options& options) = 0;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000723
deadbeef41b07982015-12-01 15:01:24 -0800724 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
725 const PeerConnectionInterface::RTCConfiguration& configuration,
726 const MediaConstraintsInterface* constraints,
kwibergd1fe2812016-04-27 06:47:29 -0700727 std::unique_ptr<cricket::PortAllocator> allocator,
Henrik Boströmd03c23b2016-06-01 11:44:18 +0200728 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
hbosd7973cc2016-05-27 06:08:53 -0700729 PeerConnectionObserver* observer) = 0;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000730
htaa2a49d92016-03-04 02:51:39 -0800731 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
732 const PeerConnectionInterface::RTCConfiguration& configuration,
kwibergd1fe2812016-04-27 06:47:29 -0700733 std::unique_ptr<cricket::PortAllocator> allocator,
Henrik Boströmd03c23b2016-06-01 11:44:18 +0200734 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
hbosd7973cc2016-05-27 06:08:53 -0700735 PeerConnectionObserver* observer) = 0;
htaa2a49d92016-03-04 02:51:39 -0800736
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000737 virtual rtc::scoped_refptr<MediaStreamInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000738 CreateLocalMediaStream(const std::string& label) = 0;
739
740 // Creates a AudioSourceInterface.
741 // |constraints| decides audio processing settings but can be NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000742 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
htaa2a49d92016-03-04 02:51:39 -0800743 const cricket::AudioOptions& options) = 0;
744 // Deprecated - use version above.
745 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000746 const MediaConstraintsInterface* constraints) = 0;
747
perkja3ede6c2016-03-08 01:27:48 +0100748 // Creates a VideoTrackSourceInterface. The new source take ownership of
htaa2a49d92016-03-04 02:51:39 -0800749 // |capturer|.
perkja3ede6c2016-03-08 01:27:48 +0100750 virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
htaa2a49d92016-03-04 02:51:39 -0800751 cricket::VideoCapturer* capturer) = 0;
752 // A video source creator that allows selection of resolution and frame rate.
753 // |constraints| decides video resolution and frame rate but can
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000754 // be NULL.
htaa2a49d92016-03-04 02:51:39 -0800755 // In the NULL case, use the version above.
perkja3ede6c2016-03-08 01:27:48 +0100756 virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000757 cricket::VideoCapturer* capturer,
758 const MediaConstraintsInterface* constraints) = 0;
759
760 // Creates a new local VideoTrack. The same |source| can be used in several
761 // tracks.
perkja3ede6c2016-03-08 01:27:48 +0100762 virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
763 const std::string& label,
764 VideoTrackSourceInterface* source) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000765
766 // Creates an new AudioTrack. At the moment |source| can be NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000767 virtual rtc::scoped_refptr<AudioTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000768 CreateAudioTrack(const std::string& label,
769 AudioSourceInterface* source) = 0;
770
wu@webrtc.orga9890802013-12-13 00:21:03 +0000771 // Starts AEC dump using existing file. Takes ownership of |file| and passes
772 // it on to VoiceEngine (via other objects) immediately, which will take
wu@webrtc.orga8910d22014-01-23 22:12:45 +0000773 // the ownerhip. If the operation fails, the file will be closed.
ivocd66b44d2016-01-15 03:06:36 -0800774 // A maximum file size in bytes can be specified. When the file size limit is
775 // reached, logging is stopped automatically. If max_size_bytes is set to a
776 // value <= 0, no limit will be used, and logging will continue until the
777 // StopAecDump function is called.
778 virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
wu@webrtc.orga9890802013-12-13 00:21:03 +0000779
ivoc797ef122015-10-22 03:25:41 -0700780 // Stops logging the AEC dump.
781 virtual void StopAecDump() = 0;
782
ivoc14d5dbe2016-07-04 07:06:55 -0700783 // This function is deprecated and will be removed when Chrome is updated to
784 // use the equivalent function on PeerConnectionInterface.
785 // TODO(ivoc) Remove after Chrome is updated.
ivocc1513ee2016-05-13 08:30:39 -0700786 virtual bool StartRtcEventLog(rtc::PlatformFile file,
787 int64_t max_size_bytes) = 0;
ivoc14d5dbe2016-07-04 07:06:55 -0700788 // This function is deprecated and will be removed when Chrome is updated to
789 // use the equivalent function on PeerConnectionInterface.
790 // TODO(ivoc) Remove after Chrome is updated.
ivoc112a3d82015-10-16 02:22:18 -0700791 virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
792
ivoc14d5dbe2016-07-04 07:06:55 -0700793 // This function is deprecated and will be removed when Chrome is updated to
794 // use the equivalent function on PeerConnectionInterface.
795 // TODO(ivoc) Remove after Chrome is updated.
ivoc112a3d82015-10-16 02:22:18 -0700796 virtual void StopRtcEventLog() = 0;
797
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000798 protected:
799 // Dtor and ctor protected as objects shouldn't be created or deleted via
800 // this interface.
801 PeerConnectionFactoryInterface() {}
802 ~PeerConnectionFactoryInterface() {} // NOLINT
803};
804
805// Create a new instance of PeerConnectionFactoryInterface.
Taylor Brandstettera8415fe2016-03-23 10:38:07 -0700806//
807// This method relies on the thread it's called on as the "signaling thread"
808// for the PeerConnectionFactory it creates.
809//
810// As such, if the current thread is not already running an rtc::Thread message
811// loop, an application using this method must eventually either call
812// rtc::Thread::Current()->Run(), or call
813// rtc::Thread::Current()->ProcessMessages() within the application's own
814// message loop.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000815rtc::scoped_refptr<PeerConnectionFactoryInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000816CreatePeerConnectionFactory();
817
818// Create a new instance of PeerConnectionFactoryInterface.
Taylor Brandstettera8415fe2016-03-23 10:38:07 -0700819//
danilchape9021a32016-05-17 01:52:02 -0700820// |network_thread|, |worker_thread| and |signaling_thread| are
821// the only mandatory parameters.
Taylor Brandstettera8415fe2016-03-23 10:38:07 -0700822//
823// If non-null, ownership of |default_adm|, |encoder_factory| and
824// |decoder_factory| are transferred to the returned factory.
danilchape9021a32016-05-17 01:52:02 -0700825rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
826 rtc::Thread* network_thread,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000827 rtc::Thread* worker_thread,
828 rtc::Thread* signaling_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000829 AudioDeviceModule* default_adm,
830 cricket::WebRtcVideoEncoderFactory* encoder_factory,
831 cricket::WebRtcVideoDecoderFactory* decoder_factory);
832
gyzhou95aa9642016-12-13 14:06:26 -0800833// Create a new instance of PeerConnectionFactoryInterface with external audio
834// mixer.
835//
836// If |audio_mixer| is null, an internal audio mixer will be created and used.
837rtc::scoped_refptr<PeerConnectionFactoryInterface>
838CreatePeerConnectionFactoryWithAudioMixer(
839 rtc::Thread* network_thread,
840 rtc::Thread* worker_thread,
841 rtc::Thread* signaling_thread,
842 AudioDeviceModule* default_adm,
843 cricket::WebRtcVideoEncoderFactory* encoder_factory,
844 cricket::WebRtcVideoDecoderFactory* decoder_factory,
845 rtc::scoped_refptr<AudioMixer> audio_mixer);
846
danilchape9021a32016-05-17 01:52:02 -0700847// Create a new instance of PeerConnectionFactoryInterface.
848// Same thread is used as worker and network thread.
danilchape9021a32016-05-17 01:52:02 -0700849inline rtc::scoped_refptr<PeerConnectionFactoryInterface>
850CreatePeerConnectionFactory(
851 rtc::Thread* worker_and_network_thread,
852 rtc::Thread* signaling_thread,
853 AudioDeviceModule* default_adm,
854 cricket::WebRtcVideoEncoderFactory* encoder_factory,
855 cricket::WebRtcVideoDecoderFactory* decoder_factory) {
856 return CreatePeerConnectionFactory(
857 worker_and_network_thread, worker_and_network_thread, signaling_thread,
858 default_adm, encoder_factory, decoder_factory);
859}
860
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000861} // namespace webrtc
862
Henrik Kjellander15583c12016-02-10 10:53:12 +0100863#endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_